msmcleod

^^^ This... well actually, not your master bus, but the master fader for your hardware output. Your hardware output fader should always be at 0.0, then adjust your master fader accordingly. It's fine to also have your master bus fader at 0.0, but you must have your hardware output fader at 0.0 for it to export at the same volume.

@Milton Sica - it's always worth running your English translations back to Portuguese just to check it makes sense. Sometimes it just needs rephrasing for the translator to do a decent job.

Do NOT do this. Running a program as administrator gives a program unrestricted access to all aspects of your machine. Any plugin will also have unrestricted access, and given that these can come from any source (esp the free ones), this is extremely risky. Sometimes it's necessary to do this temporarily for older plugins (i.e. Cakewalk Dimensions), but you should not run as administrator as a matter of course. You can disable the prompt as follows: Disable the UAC dialog box via User Account Control Settings 1. Press the Windows key, type control, and select Control Panel. 2. Select User Accounts from the list. 3. Now, click the User Accounts option. 4. Go to Change User Account Control settings. 5. Now drag the bar slider to Never notify. 6. Press the OK button to apply the selected setting. 7. Click Yes on the UAC window that opens.

If you're only using delay, Pre/Post will be the same thing. Pre/Post only comes into play when you have other processing going on - i.e. Pre delays before any other processing, Post adds delay after other processing. Using Pre and a delay only on one side of a stereo signal could radically affect how MS processing is handled.

@chamlin - not seeing any issues with that workflow here. I set an 8 measure loop, with a 4 measure punch in/out in the middle, and it's correctly recording the 4 measures every time. What might be the issue, is: "Then, without touching any settings, I hit record again, but this time, my recorded tracks are offset 7 measures forward, which results in the recording being misaligned, and 7 measures not recording" Check your "On Stop, Rewind to Now Marker" setting in TV Options. If this is unchecked, and you don't touch anything before pressing record again, then recording will start where you left off (assuming that's between the punch points). I'd recommend either: - Making sure "On Stop, Rewind to Now Marker" is checked, and ensuring the now time is at the beginning of the loop before you start recording (pressing W once or twice will ensure this). Then when you stop, it'll automatically rewind back to the beginning of the loop for you, so you can immediately press R. OR - Get into the habit of pressing W before pressing R again so you start at the beginning of the loop. If neither of these solves your problem, please post a video of what is happening ( Screen2Gif is a free and easy to use utility for grabbing screen videos). Be sure to record the whole screen, as the state of other options in the control bar may give us a clue to your setup and what could be causing the issue.

The CA2A doesn't use look-ahead buffers - it's the LP series that does.

You could use an articulation map to transform CC#1 to CC#11 or vice-versa. This will transform the CC#1 to CC#11 in real time without you having to modify the underlying MIDI events. If you want to change the MIDI events later, you can right click on the articulation and use one of the Apply options. This will change transform the MIDI data directly, and remove the articulation from the articulation lane: Note that you can have more than one transform per articulation, and a single articulation map can have up to 255 articulations, so you can build up a collection of transforms you use regularly, then simply draw the appropriate articulation in the PRV when required.

For audio clips, you've got 2 options: 1. Nudge the clip by 50ms; or 2. Put a delay effect in your FX bin. Channel Tools has the ability to delay tracks by small amounts, both in ms or samples. For MIDI clips, again 2 options: 1. Nudge the clip by 50ms; or 2. Use the Track Delay in the inspector. CbB only allows this to be specified in ticks, whereas Sonar you can specify the delay in either ticks or ms. For CbB, this means calculating the equivalent ticks value for 50ms. There are 960 ticks per quarter note, and tempo is specified in beats (quarter notes) per minute. So: 1. Multiply your tempo by 960 to give ticks per minute (e.g. 120bpm x 960 = 115,200 ticks per minute) 2. Divide by 360,000 to give ticks per millisecond (e.g. 115,200 / 360,000 = 0.32 ticks per millisecond) 3. Multiply this by the delay time in ms, e.g. 0.32 x 50ms = 16 ticks. So a delay of 16 ticks at 120bpm is 50ms.

In general, cheaper mixers with a USB interface only provide a stereo mix to your computer, so you can't record the individual channels. For multi-track recording, you need a multi-track interface. There are some mixers that are also multi-track interfaces, e.g. Tascam Model Series (Model 12, 16, 24 and 2400 ) - these also work as stand-alone mixers and recorders, so you get the choice of sending multiple tracks to your computer or recording multiple tracks to an SD card which you can then import into your DAW. Soundcraft Signature MTK series - similar to the Tascam Model series, except it only offers multiple track recording over USB to your DAW - i.e. you can't record to an SD card. DO NOT confuse the non-MTK series with the MTK series - e.g. the Signature 24 is stereo only, whereas the Signature 24 MTK is multi-track. This is reflected in the price, so if you think you're getting a bargain, double-check it's the right model. Zoom L8, L12, L20 - These offer both multi-track recording over USB to your DAW and SD card recording. The mixer is typical digital mixer setup rather than an analog mixer setup. AFAIK, all of the above mixers, with the exception of the Tascam Model 12 and Tascam Model 2400 record pre-EQ. In other words, it'll send the signal to your DAW before any EQ is added. This is because they're primarily designed for live use, where you'd record the clean signal to SD card/computer, and use EQ / Faders for the live mix. FWIW, this was the blocker for me with these products (the Model 2400 wasn't available at the time), so I opted for an Allen & Heath Mixwizard WZ3 analog mixer connected via its direct channel outs to 2 x Fostex VC8's which in turn go to an RME Digiface USB. If you don't mind not using a mixer, you any audio interface with ADAT inputs will suffice. For years I used the Focusrite 18i20 with a Behringer ADA8000 connected to it via ADAT - this gave me 8 inputs via the 18i20, and a further 8 inputs via the ADA8000. All the major manufacturers do equivalents, each with their own unique features. What you're looking for is an interface with at least 8 analog inputs as standard, and an ADAT port for additional inputs. As far as ADAT expanders go, the Behringer ADA8200 has now replaced the ADA8000 and has better pre-amps. Focusrite have their own equivalent expander in the Octopre, and there are plenty of others by other manufacturers.

Thanks - got a repro, and a fix for the next update.

Screenshot 2 looks correct to me. The envelope value at the end of the second clip is retained by adding a node of that value where the end of the second clip used to be. If you wanted the whole of the envelope to be moved, then the last node outside of the clip needs to be selected too. An easy way to do this would be to extend the end of the clip so it covers the last envelope node. Alternatively, you could CTRL + click any nodes after the end of the clip to include them. There's no screenshot 3 tho...

I've tried to repro using the steps above but cannot. Can you put together a video showing this happening from scratch? I normally use screen2gif for screen video captures. A sample project may help as well.

Doesn't the Backup/Restore Settings feature in preferences work with these interfaces? I don't have these exact ones, but the last time I used them to switch between my RME and Focusrite, it worked fine.

I believe you can load VST3 in SONAR 8 by hosting them Patchwork from Blue Cat Audio. You insert patchwork into your track as a VST2 effect, then load the VST3 within Patchwork itself.

SysEx view is there. You're probably using the Basic workspace - set it to Advanced or None.